Improving the quality of service for Voice over Internet Protocol (VoIP) over wireless Local Area Networks (LANs) is critical to achieving the vision of the unwired enterprise. Achieving high quality VoIP poses several challenges, particularly in heavily loaded systems. Voice traffic generally requires higher levels of link performance (i.e. lower delay, jitter, and frame loss) than data because voice sessions must occur in real time and transmitted frames must be received with only a small tolerance in timing or loss. Achieving high voice quality in wireless environments becomes even more challenging due to the possibility of higher noise levels and the varying signal strengths due to barriers and varying distances to the access point (AP). Changing signal strength may increase bandwidth requirements and hence affect voice quality if the bandwidth is not available.
As a calling party or mobile user moves relative to an access point, the RF bandwidth requirements may change (increase or decrease) drastically, depending on the change in distance. Since this change is not accounted for by traditional call admission control algorithms, resource overload can result. Counting sessions as they arrive, for instance, and denying calls beyond a specific number requires over-engineering to account for bandwidth variability. RF-based connections, unlike wired connections, are subject to much more interference (i.e., other radios, noise sources, etc.), which can affect RF bandwidth requirements for the session. RF bandwidth is partitioned across different traffic classes, voice, video, and data, which may result in unused RF bandwidth, if not needed by one of the traffic classes. As callers move from one access point zone to the next access point zone, the likelihood of the adjacent access point taking over the call must remain high (almost independent of call volume) to prevent unexpected disconnections.